3rd Edition, Chapter 5faculty.wiu.edu/Y-Kim2/CS556MultimediaNetwork.pdfMultmedia Networking. Multimedia: audio. Multmedia Networking. analog audio signal sampled at constant rate telephone:

Multimedia Networking

Multmedia Networking

Multimedia: audio


analog audio signal sampled at constant rate

telephone: 8,000 samples/sec

CD music: 44,100 samples/sec

each sample quantized, i.e., rounded

e.g., 28=256 possible quantized values

each quantized value represented by bits, e.g., 8 bits for 256 values

time

au

dio

sig

na

l a

mp

litu

de

analog

signal

quantized

value of

analog value

quantization

error

sampling rate

(N sample/sec)

Multimedia: audio


example: 8,000 samples/sec, 256 quantized values: 64,000 bps

receiver converts bits back to analog signal:

some quality reduction

example rates CD: 1.411 Mbps

MP3: 96, 128, 160 kbps

Internet telephony: 5.3 kbps and up

time

au

dio

sig

na

l a

mp

litu

de

analog

signal

quantized

value of

analog value

quantization

error

sampling rate

(N sample/sec)

video: sequence of images displayed at constant rate

e.g. 24 images/sec

digital image: array of pixels

each pixel represented by bits

coding: use redundancy within and between images to decrease # bits used to encode image

spatial (within image)

temporal (from one image to next)


Multimedia: video

……………………...…

spatial coding example: instead

of sending N values of same

color (all purple), send only two

values: color value (purple) and

number of repeated values (N)

……………………...…

frame i

frame i+1

temporal coding example:

instead of sending

complete frame at i+1,

send only differences from

frame i


Multimedia: video

……………………...…

spatial coding example: instead

of sending N values of same

color (all purple), send only two

values: color value (purple) and

number of repeated values (N)

……………………...…

frame i

frame i+1

temporal coding example:

instead of sending

complete frame at i+1,

send only differences from

frame i

CBR: (constant bit rate): video encoding rate fixed

VBR: (variable bit rate): video encoding rate changes as amount of spatial, temporal coding changes

examples:

MPEG 1 (CD-ROM) 1.5 Mbps

MPEG2 (DVD) 3-6 Mbps

MPEG4 (often used in Internet, < 1 Mbps)

Multimedia networking: 3 application types


streaming, stored audio, video streaming: can begin playout before downloading entire

file

stored (at server): can transmit faster than audio/video will be rendered (implies storing/buffering at client)

e.g., YouTube, Netflix, Hulu

conversational voice/video over IP interactive nature of human-to-human conversation

limits delay tolerance

e.g., Skype

streaming live audio, video e.g., live sporting event (futbol)

Streaming stored video:

1. video

recorded

(e.g., 30

frames/sec)

2. video

sent

streaming: at this time, client

playing out early part of video,

while server still sending later

part of video

network delay

(fixed in this

example)time


3. video received,

played out at client

(30 frames/sec)

Streaming stored video: challenges

continuous playout constraint: once client playout

begins, playback must match original timing

… but network delays are variable (jitter), so

will need client-side buffer to match playout

requirements

other challenges:

client interactivity: pause, fast-forward,

rewind, jump through video

video packets may be lost, retransmitted


constant bit

rate video

transmission

time

variable

network

delay

client video

receptionconstant bit

rate video

playout at client

client playout

delayb

uffe

red

vid

eo

client-side buffering and playout delay: compensate for network-added delay, delay jitter


Streaming stored video: revisted

Client-side buffering, playout


variable fill

rate, x(t)

client application

buffer, size B

playout rate,

e.g., CBR r

buffer fill level,

Q(t)

video server

client



variable fill

rate, x(t)

client application

buffer, size B

playout rate,

e.g., CBR r�

buffer fill level,

Q(t)

video server

client

1. Initial fill of buffer until playout begins at tp2. playout begins at tp,

3. buffer fill level varies over time as fill rate x(t) varies

and playout rate r is constant

playout buffering: average fill rate (x), playout rate (r):x < r: buffer eventually empties (causing freezing of video playout until buffer again fills)

x > r: buffer will not empty, provided initial playout delay is large enough to absorb variability in x(t)

initial playout delay tradeoff: buffer starvation less likely with larger delay, but larger delay until user begins watching


variable fill

rate, x(t)

client application

buffer, size B

playout rate,

e.g., CBR r

buffer fill level,

Q(t)

video server


Streaming multimedia: UDP

server sends at rate appropriate for client

often: send rate = encoding rate = constant rate

transmission rate can be oblivious to congestion levels

short playout delay (2-5 seconds) to remove network jitter

error recovery: application-level, timeipermitting

RTP [RFC 2326]: multimedia payload types

UDP may not go through firewalls


��

Streaming multimedia: HTTP

multimedia file retrieved via HTTP GET

send at maximum possible rate under TCP

fill rate fluctuates due to TCP congestion control, retransmissions (in-order delivery)

larger playout delay: smooth TCP delivery rate

HTTP/TCP passes more easily through firewalls


variable

rate, x(t) �

TCP send

buffer

video

fileTCP receive

buffer

application

playout buffer

server client

Streaming multimedia: DASH

DASH: Dynamic, Adaptive Streaming over HTTP

server: divides video file into multiple chunks

each chunk stored, encoded at different rates

manifest file: provides URLs for different chunks

client: periodically measures server-to-client bandwidth

consulting manifest, requests one chunk at a time

• chooses maximum coding rate sustainable given

current bandwidth

• can choose different coding rates at different points

in time (depending on available bandwidth at time)


Streaming multimedia: DASH

DASH: Dynamic, Adaptive Streaming over HTTP

“intelligence” at client: client determines when to request chunk (so that buffer starvation, or

overflow does not occur)

what encoding rate to request (higher quality when more bandwidth available)

where to request chunk (can request from URL server that is “close” to client or has high available bandwidth)


Content distribution networks

challenge: how to stream content (selected from millions of videos) to hundreds of thousands of simultaneous users?

option 1: single, large “mega-server” single point of failure

point of network congestion

long path to distant clients

multiple copies of video sent over outgoing link

….quite simply: this solution doesn’t scale


Content distribution networks

challenge: how to stream content (selected from millions of videos) to hundreds of thousands of simultaneous users?

option 2: store/serve multiple copies of videos at multiple geographically distributed sites (CDN) enter deep: push CDN servers deep into many access

networks

• close to users

• used by Akamai, 1700 locations

bring home: smaller number (10’s) of larger clusters in POPs near (but not within) access networks

• used by Limelight


CDN: “simple” content access scenario


Bob (client) requests video http://netcinema.com/6Y7B23V

video stored in CDN at http://KingCDN.com/NetC6y&B23V

netcinema.com

KingCDN.com

1

1. Bob gets URL for for video

http://netcinema.com/6Y7B23V

from netcinema.com

web page 2

2. resolve http://netcinema.com/6Y7B23V

via Bob’s local DNS

netcinema’s

authorative DNS

3

3. netcinema’s DNS returns URL

http://KingCDN.com/NetC6y&B23V4

4&5. Resolve

http://KingCDN.com/NetC6y&B23

via KingCDN’s authoritative DNS,

which returns IP address of KIingCDN

server with video

56. request video from

KINGCDN server,

streamed via HTTP

KingCDN

authoritative DNS

CDN cluster selection strategy

challenge: how does CDN DNS select “good” CDN node to stream to client pick CDN node geographically closest to client

pick CDN node with shortest delay (or min # hops) to client (CDN nodes periodically ping access ISPs, reporting results to CDN DNS)

IP anycast

alternative: let client decide - give client a list of several CDN servers client pings servers, picks “best”

Netflix approach


Case study: Netflix

30% downstream US traffic in 2011

owns very little infrastructure, uses 3rd party services:

own registration, payment servers

Amazon (3rd party) cloud services:

• Netflix uploads studio master to Amazon cloud

• create multiple version of movie (different

endodings) in cloud

• upload versions from cloud to CDNs

• Cloud hosts Netflix web pages for user browsing

three 3rd party CDNs host/stream Netflix content: Akamai, Limelight, Level-3


Case study: Netflix


1

1. Bob manages

Netflix account

Netflix registration,

accounting servers

Amazon cloud

Akamai CDN

Limelight CDN

Level-3 CDN

2

2. Bob browses

Netflix video3

3. Manifest file

returned for

requested video

4. DASH

streaming

upload copies of

multiple versions of

video to CDNs

Voice-over-IP (VoIP)


VoIP end-end-delay requirement: needed to maintain “conversational” aspect higher delays noticeable, impair interactivity

< 150 msec: good

> 400 msec bad

includes application-level (packetization,playout), network delays

session initialization: how does callee advertise IP address, port number, encoding algorithms?

value-added services: call forwarding, screening, recording

emergency services: 911

VoIP characteristics

speaker’s audio: alternating talk spurts, silent periods.

64 kbps during talk spurt

pkts generated only during talk spurts

20 msec chunks at 8 Kbytes/sec: 160 bytes of data

application-layer header added to each chunk

chunk+header encapsulated into UDP or TCP segment

application sends segment into socket every 20 msec during talkspurt


VoIP: packet loss, delay

network loss: IP datagram lost due to network congestion (router buffer overflow)

delay loss: IP datagram arrives too late for playout at receiver delays: processing, queueing in network; end-system

(sender, receiver) delays

typical maximum tolerable delay: 400 ms

loss tolerance: depending on voice encoding, loss concealment, packet loss rates between 1% and 10% can be tolerated


constant bit

rate

transmission

time

variable

network

delay

(jitter)

client

receptionconstant bit

rate playout

at client

client playout

delayb

uffe

red

da

ta

Delay jitter

end-to-end delays of two consecutive packets: difference can be more or less than 20 msec(transmission time difference)


VoIP: fixed playout delay

receiver attempts to playout each chunk exactly qmsecs after chunk was generated.

chunk has time stamp t: play out chunk at t+q

chunk arrives after t+q: data arrives too late for playout: data “lost”

tradeoff in choosing q:

large q: less packet loss

small q: better interactive experience


packets

time

packets

generated

packets

received

loss

r

p p'

playout schedule

p' - r

playout schedule

p - r

sender generates packets every 20 msec during talk spurt.

first packet received at time r

first playout schedule: begins at p

second playout schedule: begins at p’


VoIP: fixed playout delay

Adaptive playout delay (1)

goal: low playout delay, low late loss rate

approach: adaptive playout delay adjustment: estimate network delay, adjust playout delay at

beginning of each talk spurt

silent periods compressed and elongated

chunks still played out every 20 msec during talk spurt

adaptively estimate packet delay: (EWMA -exponentially weighted moving average, recall TCP RTT estimate):


di = (1-a)di-1 + a (ri – ti)

delay estimate

after ith packetsmall constant,

e.g. 0.1time received - time sent

(timestamp)

measured delay of ith packet

also useful to estimate average deviation of delay, vi :

estimates di, vi calculated for every received packet, but used only at start of talk spurt

for first packet in talk spurt, playout time is:

remaining packets in talkspurt are played out periodically


vi = (1-b)vi-1 + b |ri – ti – di|

playout-timei = ti + di + Kvi


Q: How does receiver determine whether packet is first in a talkspurt?

if no loss, receiver looks at successive timestamps difference of successive stamps > 20 msec -->talk spurt

begins.

with loss possible, receiver must look at both time stamps and sequence numbers difference of successive stamps > 20 msec and sequence

numbers without gaps --> talk spurt begins.



VoiP: recovery from packet loss (1)

Challenge: recover from packet loss given small tolerable delay between original transmission and playout

each ACK/NAK takes ~ one RTT

alternative: Forward Error Correction (FEC)

send enough bits to allow recovery without retransmission (recall two-dimensional parity in Ch. 5)

simple FEC for every group of n chunks, create redundant chunk by

exclusive OR-ing n original chunks

send n+1 chunks, increasing bandwidth by factor 1/n

can reconstruct original n chunks if at most one lost chunk from n+1 chunks, with playout delay


another FEC scheme:“piggyback lower

quality stream”send lower resolution

audio stream as

redundant information

e.g., nominal

stream PCM at 64 kbps

and redundant stream

GSM at 13 kbps

non-consecutive loss: receiver can conceal loss

generalization: can also append (n-1)st and (n-2)nd low-bit rate

chunk



interleaving to conceal loss: audio chunks divided into

smaller units, e.g. four 5 msec units per 20 msecaudio chunk

packet contains small units from different chunks

if packet lost, still have mostof every original chunk

no redundancy overhead, but increases playout delay




supernode overlay

network

Voice-over-IP: Skype

proprietary application-layer protocol (inferred via reverse engineering)

encrypted msgs

P2P components:

Skype clients (SC)

clients: skype peers connect directly to each other for VoIP call

super nodes (SN):skype peers with special functions

overlay network: among SNs to locate SCs

login server

Skype login server supernode (SN)


P2P voice-over-IP: skype

skype client operation:

1. joins skype network by contacting SN (IP address cached) using TCP

2. logs-in (usename, password) to centralized skype login server

3. obtains IP address for callee from SN, SN overlayor client buddy list

4. initiate call directly to callee

Skype login server


problem: both Alice, Bob are behind “NATs” NAT prevents outside peer

from initiating connection to insider peer

inside peer can initiate connection to outside

relay solution:Alice, Bob maintain open connection to their SNs Alice signals her SN to connect

to Bob Alice’s SN connects to Bob’s

SN Bob’s SN connects to Bob over

open connection Bob initially initiated to his SN

Skype: peers as relays

Real-Time Protocol (RTP)

RTP specifies packet structure for packets carrying audio, video data

RFC 3550

RTP packet provides payload type

identification

packet sequence numbering

time stamping

RTP runs in end systems

RTP packets encapsulated in UDP segments

interoperability: if two VoIP applications run RTP, they may be able to work together


RTP runs on top of UDP

RTP libraries provide transport-layer interface

that extends UDP:

• port numbers, IP addresses

• payload type identification

• packet sequence numbering

• time-stamping


RTP example

example: sending 64 kbps PCM-encoded voice over RTP

application collects encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk

audio chunk + RTP header form RTP packet, which is encapsulated in UDP segment

RTP header indicates type of audio encoding in each packet sender can change

encoding during conference

RTP header also contains sequence numbers, timestamps


RTP and QoS

RTP does not provide any mechanism to ensure timely data delivery or other QoS guarantees

RTP encapsulation only seen at end systems (notby intermediate routers)

routers provide best-effort service, making no special effort to ensure that RTP packets arrive at destination in timely matter


RTP header

payload type (7 bits): indicates type of encoding currently being

used. If sender changes encoding during call, sender

informs receiver via payload type fieldPayload type 0: PCM mu-law, 64 kbps

Payload type 3: GSM, 13 kbps

Payload type 7: LPC, 2.4 kbps

Payload type 26: Motion JPEG

Payload type 31: H.261

Payload type 33: MPEG2 video

sequence # (16 bits): increment by one for each RTP packet sent

detect packet loss, restore packet sequence


payload type

sequence number

type

time stamp SynchronizationSource ID

Miscellaneous

fields

timestamp field (32 bits long): sampling instant of first byte in this RTP data packet for audio, timestamp clock increments by one for each

sampling period (e.g., each 125 usecs for 8 KHz sampling clock)

if application generates chunks of 160 encoded samples, timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive.

SSRC field (32 bits long): identifies source of RTP stream. Each stream in RTP session has distinct SSRC


RTP header

payload type

sequence number

type

time stamp SynchronizationSource ID

Miscellaneous

fields

RTSP/RTP programming assignment

build a server that encapsulates stored video frames into RTP packets grab video frame, add RTP headers, create UDP

segments, send segments to UDP socket

include seq numbers and time stamps

client RTP provided for you

also write client side of RTSP issue play/pause commands

server RTSP provided for you


Real-Time Control Protocol (RTCP)

works in conjunction with RTP

each participant in RTP session periodically sends RTCP control packets to all other participants

each RTCP packet contains sender and/or receiver reports report statistics useful to

application: # packets sent, # packets lost, interarrival jitter

feedback used to control performance sender may modify its

transmissions based on feedback


RTCP: multiple multicast senders

each RTP session: typically a single multicast address; all RTP

/RTCP packets belonging to session use multicast address

RTP, RTCP packets distinguished from each other via distinct port

numbers

to limit traffic, each participant reduces RTCP traffic as number of

conference participants increases Multmedia Networking

RTCP

RTP

RTCPRTCP

sender

receivers

RTCP: packet types

receiver report packets: fraction of packets lost, last

sequence number, average interarrival jitter

sender report packets: SSRC of RTP stream,

current time, number of packets sent, number of bytes sent

source description packets: e-mail address of sender,

sender's name, SSRC of associated RTP stream

provide mapping between the SSRC and the user/host name


RTCP: stream synchronization

RTCP can synchronize different media streams within a RTP session

e.g., videoconferencing app: each sender generates one RTP stream for video, one for audio.

timestamps in RTP packets tied to the video, audio sampling clocks

not tied to wall-clock time

each RTCP sender-report packet contains (for most recently generated packet in associated RTP stream):

timestamp of RTP packet

wall-clock time for when packet was created

receivers uses association to synchronize playout of audio, video


RTCP: bandwidth scaling

RTCP attempts to limit its traffic to 5% of session bandwidth

example : one sender, sending video at 2 Mbps

RTCP attempts to limit RTCP traffic to 100 Kbps

RTCP gives 75% of rate to receivers; remaining 25% to sender

75 kbps is equally shared among receivers: with R receivers, each receiver

gets to send RTCP traffic at 75/R kbps.

sender gets to send RTCP traffic at 25 kbps.

participant determines RTCP packet transmission period by calculating avg RTCP packet size (across entire session) and dividing by allocated rate


SIP: Session Initiation Protocol [RFC 3261]

long-term vision:

all telephone calls, video conference calls take place over Internet

people identified by names or e-mail addresses, rather than by phone numbers

can reach callee (if callee so desires), no matter where callee roams, no matter what IP device callee is currently using


SIP services

SIP provides mechanisms for call setup:

for caller to let callee know she wants to establish a call

so caller, callee can agree on media type, encoding

to end call

determine current IP address of callee: maps mnemonic

identifier to current IP address

call management: add new media

streams during call

change encoding during call

invite others

transfer, hold calls


Example: setting up call to known IP address

Alice’s SIP invite message

indicates her port number, IP

address, encoding she prefers

to receive (PCM mlaw)

Bob’s 200 OK message

indicates his port number, IP

address, preferred encoding

(GSM)

SIP messages can be sent

over TCP or UDP; here sent

over RTP/UDP

default SIP port number is

5060time time

Bob's

terminal rings

Alice

167.180.112.24

Bob

193.64.210.89

port 5060

port 38060

m Law audio

GSMport 48753

INVITE [email protected]=IN IP4 167.180.112.24m=audio 38060 RTP/AVP 0port 5060

200 OK

c=IN IP4 193.64.210.89

m=audio 48753 RTP/AVP 3

ACKport 5060


Setting up a call (more)

codec negotiation: suppose Bob doesn’t

have PCM mlaw encoder

Bob will instead reply with 606 Not Acceptable Reply, listing his encoders. Alice can then send new INVITE message, advertising different encoder

rejecting a call Bob can reject with

replies “busy,” “gone,”“payment required,”“forbidden”

media can be sent over RTP or some other protocol


Example of SIP message

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 167.180.112.24

From: sip:[email protected]

To: sip:[email protected]

Call-ID: [email protected]

Content-Type: application/sdp

Content-Length: 885

c=IN IP4 167.180.112.24

m=audio 38060 RTP/AVP 0

Notes:

HTTP message syntax

sdp = session description protocol

Call-ID is unique for every call

Here we don’t know

Bob’s IP address

intermediate SIP

servers needed

Alice sends, receives

SIP messages using SIP

default port 506

Alice specifies in

header that SIP client

sends, receives SIP

messages over UDP


Name translation, user location

caller wants to call callee, but only has callee’s name or e-mail address.

need to get IP address of callee’s current host: user moves around

DHCP protocol

user has different IP devices (PC, smartphone, car device)

result can be based on: time of day (work,

home)

caller (don’t want boss to call you at home)

status of callee (calls sent to voicemail when callee is already talking to someone)


SIP registrar

REGISTER sip:domain.com SIP/2.0

Via: SIP/2.0/UDP 193.64.210.89

From: sip:[email protected]

To: sip:[email protected]

Expires: 3600

one function of SIP server: registrar

when Bob starts SIP client, client sends SIP REGISTER

message to Bob’s registrar server

register message:


SIP proxy

another function of SIP server: proxy

Alice sends invite message to her proxy server contains address sip:[email protected]

proxy responsible for routing SIP messages to callee, possibly through multiple proxies

Bob sends response back through same set of SIP proxies

proxy returns Bob’s SIP response message to Alice contains Bob’s IP address

SIP proxy analogous to local DNS server plus TCP setup


SIP example: [email protected] calls [email protected]


1

1. Jim sends INVITE

message to UMass

SIP proxy.

2. UMass proxy forwards request

to Poly registrar server

2 3. Poly server returns redirect response,

indicating that it should try [email protected]

5. eurecom

registrar

forwards INVITE

to 197.87.54.21,

which is running

keith’s SIP

client

5

4

4. Umass proxy forwards request

to Eurecom registrar server

86

7

6-8. SIP response returned to Jim

9

9. Data flows between clients

UMass

SIP proxy

Poly SIP

registrar

Eurecom SIP

registrar

197.87.54.21

128.119.40.186

Comparison with H.323

H.323: another signaling protocol for real-time, interactive multimedia

H.323: complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport, codecs

SIP: single component. Works with RTP, but does not mandate it. Can be combined with other protocols, services

H.323 comes from the ITU (telephony)

SIP comes from IETF: borrows much of its concepts from HTTP

SIP has Web flavor; H.323 has telephony flavor

SIP uses KISS principle: Keep It Simple Stupid


Network support for multimedia


Dimensioning best effort networks

approach: deploy enough link capacity so that congestion doesn’t occur, multimedia traffic flows without delay or loss low complexity of network mechanisms (use current “best

effort” network)

high bandwidth costs

challenges: network dimensioning: how much bandwidth is “enough?”

estimating network traffic demand: needed to determine how much bandwidth is “enough” (for that much traffic)


Providing multiple classes of service

thus far: making the best of best effort service one-size fits all service model

alternative: multiple classes of service partition traffic into classes

network treats different classes of traffic differently (analogy: VIP service versus regular service)

0111

granularity: differential

service among multiple

classes, not among

individual connections

history: ToS bits


Multiple classes of service: scenario

R1R2

H1

H2

H3

H41.5 Mbps linkR1 output

interface

queue


Scenario 1: mixed HTTP and VoIP

example: 1Mbps VoIP, HTTP share 1.5 Mbps link. HTTP bursts can congest router, cause audio loss

want to give priority to audio over HTTP

packet marking needed for router to distinguish

between different classes; and new router policy to

treat packets accordingly

Principle 1

R1R2


Principles for QOS guarantees (more)

what if applications misbehave (VoIP sends higher than declared rate) policing: force source adherence to bandwidth allocations

marking, policing at network edge

provide protection (isolation) for one class from others

Principle 2

R1 R2

1.5 Mbps link

1 Mbps

phone

packet marking and policing


allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use its allocation

while providing isolation, it is desirable to use

resources as efficiently as possible

Principle 3

R1R2

1.5 Mbps link

1 Mbps

phone

1 Mbps logical link

0.5 Mbps logical link


Principles for QOS guarantees (more)

Scheduling and policing mechanisms

scheduling: choose next packet to send on link

FIFO (first in first out) scheduling: send in order of arrival to queue real-world example?

discard policy: if packet arrives to full queue: who to discard?

• tail drop: drop arriving packet

• priority: drop/remove on priority basis

• random: drop/remove randomly


queue

(waiting area)

packet

arrivalspacket

departureslink

(server)

Scheduling policies: priority

priority scheduling: send highest priority queued packet

multiple classes, with different priorities class may depend on

marking or other header info, e.g. IP source/dest, port numbers, etc.

real world example?


high priority queue

(waiting area)

low priority queue

(waiting area)

arrivals

classify

departures

link

(server)

1 3 2 4 5

5

5

2

2

1

1

3

3 4

4

arrivals

departures

packet in

service

Scheduling policies: still more

Round Robin (RR) scheduling:

multiple classes

cyclically scan class queues, sending one complete packet from each class (if available)

real world example?


1 23 4 5

5

5

2

3

1

1

3

3 4

4

arrivals

departures

packet in

service

Weighted Fair Queuing (WFQ):

generalized Round Robin

each class gets weighted amount of service in each cycle

real-world example?


Scheduling policies: still more

Policing mechanisms

goal: limit traffic to not exceed declared parameters

Three common-used criteria:

(long term) average rate: how many pkts can be sent per unit time (in the long run) crucial question: what is the interval length: 100 packets

per sec or 6000 packets per min have same average!

peak rate: e.g., 6000 pkts per min (ppm) avg.; 1500 ppm peak rate

(max.) burst size: max number of pkts sent consecutively (with no intervening idle)


Policing mechanisms: implementation

token bucket: limit input to specified burst size and average rate

bucket can hold b tokens

tokens generated at rate r token/sec unless bucket full

over interval of length t: number of packets admitted less than or equal to (r t + b)


Policing and QoS guarantees

token bucket, WFQ combine to provide guaranteed upper bound on delay, i.e., QoS guarantee!

WFQ

token rate, r

bucket size, b

per-flow

rate, R

D = b/Rmax

arriving

traffic


arriving

traffic

Differentiated services

want “qualitative” service classes “behaves like a wire” relative service distinction: Platinum, Gold, Silver

scalability: simple functions in network core, relatively complex functions at edge routers (or hosts) signaling, maintaining per-flow router state difficult

with large number of flows

don’t define define service classes, provide functional components to build service classes


edge router:

per-flow traffic management

marks packets as in-profile and

out-profile

core router:

per class traffic management

buffering and scheduling based

on marking at edge

preference given to in-profile

packets over out-of-profile

packets

Diffserv architecture


r

b

marking

scheduling

...

Edge-router packet marking

class-based marking: packets of different classes marked

differently

intra-class marking: conforming portion of flow marked

differently than non-conforming one

profile: pre-negotiated rate r, bucket size b

packet marking at edge based on per-flow profile

possible use of marking:

user packets

rate r

b


Diffserv packet marking: details

packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6

6 bits used for Differentiated Service Code Point (DSCP) determine PHB that the packet will receive

2 bits currently unused


DSCP unused

Classification, conditioning

may be desirable to limit traffic injection rate of some class:

user declares traffic profile (e.g., rate, burst size)

traffic metered, shaped if non-conforming


Forwarding Per-hop Behavior (PHB)

PHB result in a different observable (measurable) forwarding performance behavior

PHB does not specify what mechanisms to use to ensure required PHB performance behavior

examples: class A gets x% of outgoing link bandwidth over time

intervals of a specified length

class A packets leave first before packets from class B


Forwarding PHB

PHBs proposed:

expedited forwarding: pkt departure rate of a class equals or exceeds specified rate logical link with a minimum guaranteed rate

assured forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth

each with three drop preference partitions


Per-connection QOS guarantees

basic fact of life: can not support traffic demands beyond link capacity

call admission: flow declares its needs, network may

block call (e.g., busy signal) if it cannot meet needs

Principle 4

R1R2

1.5 Mbps link

1 Mbps

phone

1 Mbps

phone


QoS guarantee scenario

resource reservation call setup, signaling (RSVP)

traffic, QoS declaration

per-element admission control

QoS-sensitive scheduling

(e.g., WFQ)

request/

reply


Documents

3rd Edition, Chapter 5faculty.wiu.edu/Y-Kim2/CS556MultimediaNetwork.pdfMultmedia Networking. Multimedia: audio. Multmedia Networking. analog audio signal sampled at constant rate telephone: